Apparatus, method and electroacoustic system for reverberation time extension

ABSTRACT

An apparatus for reverberation time extension is provided. The apparatus includes a module for calculating wave field synthesis information, and a signal processor for generating a plurality of audio output signals for a plurality of loudspeakers based on the audio input signals that have been recorded by a plurality of microphones, and based on the wave field synthesis information. Further, the apparatus includes an operating unit for determining a virtual position of one or several virtual walls. The module for calculating wave field synthesis information is implemented to calculate the wave field synthesis information based on the virtual position of the one or several virtual walls. Further, the virtual position is adjustable by the operating unit for at least one of the virtual walls.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation of copending InternationalApplication No. PCT/EP2012/066392, filed Aug. 23, 2012, which isincorporated herein by reference in its entirety, and additionallyclaims priority from German Application No. 102011082310.7, filed Sep.7, 2011, and U.S. Application 61/531,899, filed Sep. 7, 2011, which areall incorporated herein by reference in their entirety.

BACKGROUND OF THE INVENTION

The present invention relates to an apparatus, a method and anelectroacoustic system for reverberation time extension.

From an acoustic point of view, a room might not be optimum fordifferent applications. Thus, a musical performance normallynecessitates some reverberation to sound good. On the other hand,speakers may partly not be understood when the room is too reverberant.Thus, an adaptation of the reverberation time by means of reverberationsystems is useful.

For example, in theaters, conference centers, planetariums, seminarrooms, multifunctional rooms, different acoustic conditions may beneeded for different situations, and in particular differentrequirements with regard to reverberation time are necessitated. Forinfluencing the reverberation time, electroacoustic systems forreverberation time extension can be used. Such systems can either beincorporated into an already existing concert hall, however, it can alsobe useful to provide an electroacoustic system for reverberation timeextension already when constructing and building respective buildingsand halls, for example in exhibition construction. Reverberation timeextension can also be desirable for audio reproduction for entertainmentpurposes.

In the following, the wave field synthesis technology will be discussedin more detail. The wave field synthesis (WFS) has been researched at TUDelft and presented for the first time in the late 80's (Berkhout, A.J.; de Vries, D.; Vogel, P.: Acoustic control by Wave-field Synthesis.JASA 93, 1993).

Due to the enormous requirements of this method with regard to computingpower and transmission rates, wave field synthesis has so far beenhardly applied in practice. Only the progress in the field ofmicroprocessor technology and audio encoding allow today the usage ofthis technology in specific applications. First products in theprofessional area are expected next year. The basic idea of WFS is basedon the application of the Huygen Principle of wave theory:

Each point captured by a wave is the starting point of an elementarywave spreading spherically or circularly. Applied to acoustics, any formof an incoming wavefront can be reproduced by a large number ofloudspeakers arranged next to each other (in a so-called loudspeakerarray). In the simplest case of an individual point source to bereproduced and a linear array of loudspeakers, the audio signals of eachloudspeaker have to be fed with a time delay and amplitude scaling suchthat the radiated sound fields of the individual loudspeaker aresuperimposed properly. For several sound sources, the contribution toeach loudspeaker is calculated separately for each source and theresulting signals are added. In a room having reflective walls,reflections can also be reproduced as additional sources via theloudspeaker array. Thus, the calculation effort depends heavily on thenumber of sound sources, the reflection characteristics of the recordingroom and the number of loudspeakers.

The advantage of this technology is in particular that a natural spatialsound impression is possible across a large area of the reproductionroom. In contrast to the known technologies, direction and distance ofsound sources are reproduced very exactly. To a limited extent, virtualsound sources can even be positioned between the real loudspeaker arrayand the listener.

Thus, by the technology of wave field synthesis (WFS), a good spatialsound can be obtained for a large range of listeners. As has been statedabove, the wave field synthesis is based on the principle of Huygens,according to which wavefronts can be formed and built up bysuperimposing elementary waves. According to a mathematically exacttheoretical description, an infinite amount of sources at an infinitesmall distance would have to be used for generating the elementarywaves. Practically, however, a finite amount of loudspeakers are used ata finite small distance to each other. Each of these loudspeakers iscontrolled according to the WFS principle with an audio signal from avirtual source having a specific delay and a specific level. Level anddelays are normally different for all loudspeakers.

As has been stated above, a wave field synthesis system operates basedon the Huygen Principle and reconstructs a given waveform of, forexample, a virtual source arranged at a specific distance to a listenerby a plurality of individual waves. Thus, the wave field synthesisalgorithm receives information on the actual position of an individualloudspeaker from the loudspeaker array to calculate then a componentsignal for this individual loudspeaker, which this loudspeaker thenfinally has to radiate, so that for the listener a superposition of theloudspeaker signal from the individual loudspeaker with the loudspeakersignals of the other active loudspeakers performs a reconstruction suchthat the listener has the impression that he is not exposed to soundfrom many individual loudspeakers but merely from a single loudspeakerat the position of the virtual source.

For several virtual sources in a wave field synthesis setting, thecontribution of each virtual source for each loudspeaker, i.e. thecomponent signal of the first virtual source for the first loudspeaker,the second virtual source for the first loudspeaker, etc. is calculated,to then add up the component signals to finally obtain the actualloudspeaker signal. In the case of, for example, three virtual sources,the superposition of the loudspeaker signals of all active loudspeakerwould have the effect for the listener that the listener does not havethe impression that he is exposed to sound from a large array ofloudspeakers, but that the sound that he hears merely originates fromthree sound sources positioned at specific positions, which are equal tothe virtual sources.

In practice, calculation of the component signals is performed mostly inthat an audio signal assigned to a virtual source is provided at aspecific time with a delay and a scaling factor, depending on theposition of the virtual source and the position of the loudspeaker, inorder to obtain a delayed and/or scaled audio signal of the virtualsource, which represents the loudspeaker signal immediately when onlyone virtual source exists, or which, after addition with furthercomponent signals for the considered loudspeaker from other virtualsources, contributes to the loudspeaker signal for the consideredloudspeaker.

Typical wave field synthesis algorithms operate independent of how manyloudspeakers exist in the loudspeaker array. The theory underlying wavefield synthesis is that each arbitrary sound field can be exactlyreconstructed by an infinitely high number of individual loudspeakers,wherein individual loudspeakers are arranged infinitely close to oneanother. In practice, however, neither the infinitely high number northe infinitely close arrangement can be realized. Instead, a limitednumber of loudspeakers exist, which are additionally arranged atspecific predetermined intervals to one another. Thereby, in realsystems only an approximation of the actual waveform is obtained, whichwould take place if the virtual source actually existed, i.e. were areal source.

Wave field synthesis means are further able to reproduce severaldifferent source types. A prominent source type is the point sourcewhere the level decreases proportionally 1/r, wherein r is the distancebetween a listener and the position of the virtual source. Anothersource type is a source radiating plane waves. Here, the level remainsconstant independent of the distance to the listener, since plane wavescan be generated by point sources that are arranged at an infinitedistance to each other.

After the above excursion on existing wave field synthesis means, wewill now deal with systems for reverberation time extension known fromconventional technology:

In US005109419A, Griesinger describes an electroacoustic system forreverberation time extension where different sound sources are recordedvia microphone or direct input and are artificially reverberated via areverb matrix. The output signals of this system are output todistributed loudspeakers and thus generate an artificial reverberationin the room.

Also, Poletti describes in “Reverberators for use in wide band assistedreverberation systems” US000000039189E an electroacoustic system forreverberation time extension based on the detection of spatial signals,and processing the same in a delay matrix which again controls aplurality of loudspeakers.

In US0051425869A, Berkhout describes an approach where a signal recordedin a room is convolved and reproduced via a reconstructed wave field.

In patent literature, there are different systems for reverberation timeextension, such as U.S. Pat. No. 3,614,320 A and WO 2006092995 A1.

However, none of the systems allow flexible dynamic adaptation todifferent and alternating acoustic conditions and desires of the usersconcerning reverberation time extension.

SUMMARY

According to an embodiment, an apparatus for reverberation timeextension may have: a module for calculating wave field synthesisinformation, a signal processor for generating a plurality of audiooutput signals for a plurality of loudspeakers based on a plurality ofaudio input signals that have been recorded by a plurality ofmicrophones, and based on the wave field synthesis information, and anoperating unit for determining a virtual position of one or severalvirtual walls, wherein the module for calculating wave field synthesisinformation is implemented to calculate the wave field synthesisinformation based on the virtual position of the one or several virtualwalls, and wherein, for at least one of the virtual walls, the virtualposition is adjustable by the operating unit.

According to another embodiment, a method for reverberation timeextension may have the steps of: determining a virtual position of oneor several virtual walls; receiving a plurality of audio input signalsthat have been recorded by a plurality of microphones, calculating wavefield synthesis information, and generating a plurality of outputsignals for a plurality of loudspeakers based on the audio input signalsand based on the wave field synthesis information, wherein the wavefield synthesis information is calculated based on the virtual positionof the one or several virtual walls, and wherein the virtual position isadjustable for at least one of the virtual walls.

Another embodiment may have a computer program having a program code forperforming the inventive method, when the computer program runs on acomputer.

According to another embodiment, an electroacoustic system forreverberation time extension may have: a plurality of microphones; aninventive apparatus for reverberation time extension, and a loudspeakerarray having a plurality of loudspeakers, wherein the plurality ofmicrophones is implemented to generate a plurality of audio inputsignals fed into the apparatus for reverberation time extension, andwherein the plurality of loudspeakers of the loudspeaker array areimplemented to have the audio output signals fed in by the apparatus forreverberation time extension and to reproduce the fed-in audio outputsignals.

According to another embodiment, a method for reverberation timeextension by means of an electroacoustic system may have the steps of:recording a plurality of audio input signals by a plurality ofmicrophones; performing the inventive method for reverberation timeextension for generating a plurality of audio output signals, whereinthe step of receiving the plurality of audio input signals includes thatthat plurality of audio input signals that has been recorded by theplurality of microphones is received, and outputting the plurality ofaudio output signals by means of a loudspeaker array having a pluralityof loudspeakers.

The invention provides an apparatus for reverberation time extension.The apparatus comprises a module for calculating wave field synthesisinformation and a signal processor for generating a plurality of audiooutput signals for a plurality of loudspeakers based on a plurality ofaudio input signals, and based on the wave field synthesis information,wherein the audio signals have been recorded or captured by a pluralityof microphones. Further, the apparatus comprises an operating unit fordetermining a virtual position of one or several virtual walls. Themodule for calculating wave field synthesis information is implementedto calculate the wave field synthesis information based on the virtualposition of the one or several virtual walls. Further, the virtualposition of at least one of the virtual walls is adjustable by theoperating unit.

By shifting the virtual walls towards the outside, an acoustic roomexpansion is obtained. The acoustic room expansion is additionallyobtained by a regenerative effect which consists of recording thegenerated audio output signals output by the loudspeakers again by themicrophones and hence the same are incorporated into the generation ofthe audio output signals at a later time.

Thus, an apparatus and a method for generating an acoustic roomexpansion is provided, wherein distributed microphones record relevantsound sources and the acoustic environment and reproduce this withrespect to stationary or dynamical virtual source positions via a wavefield synthesis system.

The invention is based on the concept that the virtual sources aregenerated in an algorithm based on wave field synthesis. Here, theinvention describes a method wherein, by means of distributedmicrophones in the room to be reverberated, the acoustics of the room isrecorded with the sources to be amplified and supplied to a processingsystem via AD converters. Here, the processing system can consist of asoftware where the signal is first processed via filters and thenprocessed in a wave field synthesis algorithm to an object based soundsource, which is again processed via filters to be then output via awave field synthesis system. Due to the options of wave field synthesis,the recorded spatial signals can be positioned in any manner and can beshifted as “virtual walls” in an arbitrary manner. Thereby, individualroom geometries can be generated. The recorded spatial signals aretypically represented as plane waves and hence correspond to theacoustic effect of a wall. This virtual wall cannot only be shifted butalso be amended with respect to its angle and hence influences thereflection patterns of the sound sources directly.

In one embodiment, the module for calculating wave field synthesisinformation is implemented to calculate delay values and amplitudefactor values as wave field synthesis information. Here, the delay valuestates the delay by which one of the audio input signals is reproducedat one of the loudspeakers in a delayed manner. The amplitude factorvalue states by what factor the amplitude of one of the audio inputsignals is modified to obtain a modified signal which is output at oneof the loudspeakers.

Further, the module for calculating wave field synthesis information canbe implemented to calculate, for any point in time, a delay value and anamplitude factor value for each loudspeaker/virtual wall pair, wherein aloudspeaker/virtual wall pair is a pair of a loudspeaker and one of thevirtual walls.

In a further embodiment, the module for calculating wave field synthesisinformation is implemented to calculate the delay value and theamplitude factor value for a loudspeaker/virtual wall pair based on thedistance of the loudspeaker and the virtual wall of theloudspeaker/virtual wall pair.

Further, the module for calculating the wave field synthesis informationcan be implemented to set the delay value of a loudspeaker/virtual wallpair the higher the greater the distance between the loudspeaker and thevirtual wall is.

Further, the module for calculating wave field synthesis information canbe implemented to set the amplitude factor value of aloudspeaker/virtual wall pair the smaller the greater the distancebetween the loudspeaker and the virtual wall is.

In a further embodiment, the operating unit is implemented to shift atleast one of the virtual walls from a first virtual position to a secondvirtual position, such that the virtual wall can be shifted arbitrarilyin parallel to its first position. Further, the operating unit can beimplemented to shift at least one of the virtual walls from a firstvirtual position to a second virtual position, such that the virtualwall can be shifted arbitrarily in rotatable manner with respect to itsfirst position.

In a further embodiment, the operating unit is implemented so that thevirtual position is adjustable by the operating unit for all of thevirtual walls. The operating unit can be implemented such that each ofthe virtual walls can be shifted from a first virtual position to asecond virtual position, such that each virtual wall can be shiftedarbitrarily in parallel and in a rotatable manner with respect to itsfirst position.

In a further embodiment the apparatus for reverberation time extensioncan comprise a parametric filter for filtering resonance frequencies.

Further, an electroacoustic system for reverberation time extension isprovided including a plurality of microphones, an apparatus forreverberation time extension according to one of the above describedembodiments and a loudspeaker array of a plurality of loudspeakers.Here, the plurality of microphones is implemented to generate aplurality of audio input signals fed into the apparatus forreverberation time extension, and wherein the plurality of loudspeakersof the loudspeaker array are implemented to have the audio signals fedin by the apparatus for reverberation time extension and to reproducethe fed-in audio output signals.

BRIEF DESCRIPTION OF THE DRAWINGS

Embodiments of the present invention will be detailed subsequentlyreferring to the appended drawings, in which:

FIG. 1 illustrates a block diagram of an apparatus for reverberationtime extension according to an embodiment;

FIG. 2 illustrates a block diagram showing the cooperation of a modulefor calculating wave field synthesis information and a signal processor;

FIG. 3 illustrates an electroacoustic WFS system for reverberation timeextension according to an embodiment;

FIG. 4 illustrates a further embodiment of an electroacoustic WFSsystem;

FIG. 5 illustrates an average conference room (5 m×18 m×15 m) providedwith five ceiling microphones, 40 ceiling loudspeakers and acircumferential horizontal strip of conventional loudspeakers in areduced WFS array according to an embodiment;

FIG. 6 shows an array of loudspeaker, virtual walls and microphonesaccording to an embodiment; and

FIG. 7 shows an array of loudspeakers and a virtual wall according to afurther embodiment.

DETAILED DESCRIPTION OF THE INVENTION

FIG. 1 shows an apparatus for reverberation time extension according toan embodiment. The apparatus comprises a module for calculating wavefield synthesis information 110. Further, the apparatus comprises asignal processor 120 for generating a plurality of audio output signalsy₁, y₂, . . . , y_(n) for a plurality of loudspeakers (not shown) basedon a plurality of audio input signals s₁, s₂, . . . , s_(n) that havebeen recorded by a plurality of microphones (not shown), and based onthe wave field synthesis information. Further, the apparatus comprisesan operating unit 130 for determining a virtual position of one orseveral virtual walls. Here, the module for calculating wave fieldsynthesis information 110 is implemented to calculate the wave fieldsynthesis information WS_(inf) based on the virtual position of the oneor several virtual walls. Here, for at least one of the virtual walls,its virtual position is adjustable by the operating unit. In oneembodiment, the module for calculating wave field synthesis information110 and the signal processor 120 can be realized in one module (a wavefield synthesis module).

In the following, with reference to FIG. 2, an implementation of thecooperation of the module for calculating wave field synthesisinformation and the signal processor will be illustrated. In FIG. 2, themodule for calculating wave field synthesis 210 and the signal processor220 are illustrated by dotted lines.

The module for calculating wave field synthesis information 210 and thesignal processor 202 have a strongly parallel structure in that startingfrom the audio signal supplied to the signal processor for each virtualwall (a virtual source) and starting from the position information forthe respective virtual wall (virtual source), which the module forcalculating wave field synthesis information 210 has received from anoperating unit, first, delay information V_(i) as well as amplitudefactors (scaling factors) SF_(i) are calculated, which depend on theposition information and the position of the just consideredloudspeaker, i.e. the loudspeaker with the ordinal number j, i.e.LS_(j). Calculating delay information V_(i) as well as a scaling factorSF_(i) based on the position information of a virtual source (virtualwall) and the position of the considered loudspeaker j takes place byknown algorithms that are implemented in means 300, 302, 304, 306.

Based on the delay information V_(i)(t) and SF_(i)(t) as well as basedon the audio signal AS_(i)(t) assigned to the individual virtual source,a discrete value AW_(i)(t_(A)) for a current time t_(A) is calculatedfor the component signal K_(ij) in a finally obtained loudspeakersignal. This is performed by means 310, 312, 314, 316 as shownschematically in FIG. 2. FIG. 2 further shows actually a “flash shot” atthe time t_(A) for the individual component signals. The individualcomponent signals are then combined to nodes by a summator 320, in orderto determine the discrete value for the current time t_(A) of theloudspeaker signal for the loudspeaker j which can then be supplied tothe loudspeaker for outputting.

As can be seen in FIG. 2, first, a value valid at the current time iscalculated for each virtual source individually based on delayinformation and a scaling with an amplitude factor (scaling factor),whereupon all component signals for one loudspeaker are summed based onthe different virtual walls (virtual sources). If, for example, only onevirtual source existed, the summator would be omitted, and the signalapplied at the output of the summator in FIG. 2 would, for example,correspond to the signal output by means 310 when the virtual source 1is the only virtual source.

Here, it should be noted that the value of a loudspeaker signal isobtained at the respective output, which is a superposition of thecomponent signals for this loudspeaker due to the different virtualsources 1, 2, 3, . . . , n. Such an array would basically be providedfor each loudspeaker, except that for example 2, 4 or 8 combinedloudspeakers are controlled by the same loudspeaker signal, which isadvantageous for practical reasons.

FIG. 3 represents an electroacoustic WFS system for reverberation timeextension according to an embodiment.

According to the embodiment of FIG. 3, four microphones 350 are evenlyinstalled in a room, suspended from the ceiling by one meter. Themicrophone signals are processed to one virtual source in a WFSalgorithm 360, which is reproduced in the same room as plane wave via aWFS reverberation system 370. The WFS system includes an operatingsurface for moving the four microphone sources. With this operationunit, the four microphone signals are recorded and pulled towards theoutside. The result is an acoustic extension of the room.

In that way, rooms of any size can be generated. By positioning thevirtual walls, the reverberation time of the room changes with respectto the position and arrangement (angle) of the walls.

In one embodiment, the wave field synthesis module 360 includes a modulefor calculating wave field synthesis information according to theembodiment of FIG. 1 and a signal processor according to the embodimentof FIG. 1.

In one embodiment, the operating unit 375 is an operating unit accordingto the embodiment of FIG. 1.

According to an embodiment, unit 355 in FIG. 3 represents a filter thatserves to filter resonance frequencies. A sound wave output at one ofthe loudspeakers 370 is again recorded by the microphones 350 andconsidered again when generating the later audio signal output via theloudspeakers. In order to avoid occurring undesired resonances, thefilter 355 can be used for suppressing these resonances.

According to one embodiment, the filter 365 can be a conventional filterthat serves, for example, to adapt the loudspeakers.

FIG. 4 illustrates a further embodiment of an electroacoustic WFSsystem. The signals of the ceiling microphones are processed in acentral processing unit and processed to virtual sources after filteringin a matrix, which is again reproduced after level adaptation,regulation of the room proportion and the loudspeaker filtering asvirtual sound sources via a WFS array and equally distributed ceilingloudspeakers.

In FIG. 4, microphones 411, 412 feed audio input signals into microphonepreamplifiers 416, 417. The microphone 411 is a microphone which isclose to a sound source, e.g. a lectern. The microphone 412 is a roommicrophone which is within the room but more distant from the soundsource than the microphone 411. Normally, several room microphonesand/or several microphones close to the sound source are used.

Microphone preamplifiers 416, 417 amplify the audio input signalsreceived from the microphones 411, 412 to obtain preamplified audioinput signals. The microphone preamplifiers 416, 417 can be commonmicrophone preamplifiers. The preamplified audio input signals are fedinto an analog digital converter 420 which converts the audio signals,which are first in analog form, into digital audio signals. The analogdigital converter 420 can be a common analog digital converter.

The analog digital converter 420 feeds the digital audio signals into anabsorption filter 425. The absorption filter 425 performs filtering,which serves to adapt the same to the wall material. In one embodiment,the absorption filter 425 filters such that, when heavily reflectingwalls are to be reproduced, the digital audio signals pass theabsorption filter 425 in an almost unfiltered manner. If, however,heavily attenuating walls are to be reproduced, in one embodiment, theabsorption filter 425 filters the digital audio signal to a greatextent.

Filter 430 is a filter for feedback compensation and tone control. If aloudspeaker speaker reproduces a signal, the sound waves of this signalare again recorded by the microphone and this results in a feedback. Inone embodiment, the filter 430 can be used to compensate this feedbackcompletely or partly. Additionally, the filter 430 can be used for tonecontrol. In one embodiment, the feedback compensation and/or the tonecontrol can be performed in a conventional manner.

Further, the system in FIG. 4 includes a central operating unit 435 anda module for calculating wave field synthesis information 440. Here, thecentral operating unit 435 can correspond to the operating unit inFIG. 1. The central operating unit in FIG. 4 can be provided with a GUI(Graphical User Interface). The module for calculating wave fieldsynthesis information 440 can correspond to the module for calculatingwave field synthesis information of FIG. 1.

The module for calculating wave field synthesis information 440 passesthe calculated wave field synthesis parameter on to the module 445.These wave field synthesis parameters can, for example, be delay valuesand amplitude values, such as amplitude factor values.

The module 445 builds a delay amplitude matrix from the values passed onby module 440. In one embodiment, the delay amplitude matrix caninclude, for example, one delay value and one amplitude factor for aspecific time for each loudspeaker/virtual wall pair.

The module 445 performs audio scaling based on the wave field synthesisparameters obtained from the module for calculating wave field synthesisinformation 440. If, for example, a delay value and an amplitude factorvalue have been obtained for a loudspeaker/virtual wall pair, forexample the signal radiated by the virtual wall (e.g. seeminglyreflected by the virtual wall) is delayed by the obtained delay value,and the amplitude factor value obtained from the module 440 is modifiedby the amplitude factor value to the amplitude of the signal to beoutput, for example by multiplying the amplitude factor value with theamplitude of the signal to be output.

In the following, the filter 450 filters the audio signals modified bythe module 445 to obtain loudspeaker adaptation. In a master gain module455, the audio signals are modified to adjust the total volume. This cantake place in a conventional manner. In a gain room proportion module460, adjustment of the ratio room proportion to original signal isperformed. In one embodiment, for example, the ratio of audio signalsgenerated from audio signals of room microphones to audio signalsgenerated from audio signals of microphones close to the lectern isadjustable, for example by adjusting the amplitudes of the respectivesignals.

The modified digital audio signals are then fed into a digital analogconverter 465 that converts the modified digital audio signals intoanalog audio output signals. The analog audio output signals are thenamplified by power amplifiers 471, 472 and output by loudspeakers 481,482. In the embodiment of FIG. 4, the audio signals are output either byWFS loudspeakers 481 or ceiling loudspeakers 482. It is obvious that aplurality of WFS loudspeakers and/or ceiling loudspeakers can be used ina real system.

FIG. 5 shows an average conference room (5 m×18 m×15 m) provided with 5ceiling microphones 511, 512, 513, 514, 515, 40 ceiling loudspeakers anda circumferential horizontal strip of conventional loudspeakers in areduced WFS array 530. The signals of the ceiling microphones 511, 512,513, 514, 515 are processed in a central processing unit and processedinto virtual sources in a matrix after filtering, which is output againafter level adaptation, regulation of the room proportion andloudspeaker filtering as virtual sound sources 521, 522, 523, 524, 525via a WFS array and equally distributed ceiling loudspeakers. FIG. 4shows the structure. Here, the microphone signals are represented by therespectively opposing virtual sources to prevent feedbacks. By using aflexible matrix and the option of representing any inputs (microphoneinputs/line In) as virtual sound sources 521, 522, 523, 524, 525,directly microphoned signals are incorporated into the room simulationand due to their positioning they are also used for representation in anartificially reverberation extended room. However, these signals have tobe considered as only hardly regenerative, since they barely includeroom proportions. Further microphones can be added to record a complexdistribution of reflections. Also, the representation of virtual soundsources 521, 522, 523, 524, 525 on the ceiling is possible, which can beconsidered as significant quality requirement when representing a realroom. In the input branch of the matrix, there is, apart from a filterunit with narrow band filters for feedback suppression, also a filterunit considering different room materials in order to incorporatedifferent absorption and reflection parameters in the room to bereverberated. The detected microphone signals are, as described, againmodelled into freely positionable sources and, provided with theexisting room characteristics, again recorded by the microphone whichresults in a regeneration of the room acoustics.

FIG. 6 illustrates a basic concept of specific embodiments. Aloudspeaker array comprising, as shown in FIG. 6, 12 loudspeakers 611,612, 613 is illustrated. In real embodiments, the number of loudspeakerswill frequently be significantly higher and comprises, for example, 60,100, 200, 300 or more loudspeakers. Additionally, four virtual walls621, 622, 623, 624 are illustrated.

In the following, the loudspeaker 611 and the virtual wall 621 will beconsidered in more detail. The same form a loudspeaker/virtual wall pair(611, 621). Any other combination of one of the loudspeakers and one ofthe virtual walls also forms a loudspeaker/virtual wall pair. Thedistance between the loudspeaker and the virtual wall is indicated by anarrow d. In FIG. 6, additionally, a plurality of microphones 631, 632,633 are provided. For simplicity reasons, it is assumed that amicrophone 631 produces an audio signal by recording sound waves, whichis to be reproduced via the loudspeaker 611. Here, the signal reproducedby the loudspeaker 611 is to correspond to a reflection of the soundwaves recorded by the microphone 631 at the virtual wall 621. This meansthat the signal recorded by the microphone can only be reproduced with atime delay by the loudspeaker 611, which depends on the distance betweenloudspeaker and virtual wall: the higher the distance between virtualwall 621 and loudspeaker 611, the greater the time delay, i.e. the delayvalue by which the signal recorded by the microphone 631 is to bereproduced at the loudspeaker 611. FIG. 6 shows a shift of the virtualwall 621 by the dotted line 629, wherein the distance of the virtualwall to the loudspeaker 611 increases from d to 2 d. The delay valueincreases accordingly.

In a specific embodiment, the delay value can be calculated according tothe formula:Delay=(d+c)*p ₁wherein d is the distance between the loudspeaker and the virtual wallof the loudspeaker/virtual wall pair, c is a constant value and p₁ aproportionality constant greater than 0. The delay value becomes thegreater the greater the distance between loudspeaker and virtual wallis.

The higher the distance between virtual wall and loudspeaker becomes,the smaller, in one embodiment, the amplitude factor is to be selectedsince also the amplitude of a real sound source becomes the smaller thefurther away one is from a sound source, wherein the virtual soundsource represents the virtual wall that seemingly reflects a sound wave.The amplitude factor is the factor by which the amplitude of one of theoutput signals is to be modified to obtain a modified signal which is tobe output at one of the loudspeakers.

In a specific embodiment, the amplitude factor can be calculatedaccording to the formula:amplitude factor=[1/(d+h)]*p ₂wherein d is the distance between loudspeaker and virtual wall of theloudspeaker/virtual wall pair, h a constant value and p₂ aproportionality constant greater than 0. In embodiments, theproportionality constant p₂ is selected such that the amplitude factorassumes a value greater than 0 and less than 1.

Basically, increasing the delay value can cause reverberation timeextension.

FIG. 7 shows a further embodiment where the current position 729 of thevirtual wall is changed such that the current position 729 of thevirtual wall has been changed in a rotatable manner with respect to itsold position 721. The distance of the old position of the virtual wallto the loudspeaker 711 is illustrated by arrow e, the distance of thenew position of the virtual wall to the loudspeaker 711 is illustratedby arrow f.

While some aspects have been described in the context of an apparatus,it is obvious that these aspects also represent a description of therespective method, wherein a block or an apparatus corresponds to amethod step or a feature of a method step. Analogously, aspects thathave been described in the context of a method step represent also adescription of a respective block or element or feature of a respectiveapparatus.

An inventive computer program or signal can be stored in a digitalmemory medium or can be transferred on a transfer medium, such as awireless transfer medium or a wired transfer medium, such as theInternet.

Depending on specific implementation requirements, embodiments of theinvention can be implemented in hardware or in software. Theimplementation can be made by using a digital memory medium, such as adisc, a DVD, a CD, an ROM, a PROM, an EPROM, an EEPROM or a Flash memoryon which electronically readable control signals are stored, whichcooperate (or are able to cooperate) with a programmable computer systemsuch that the respective method is performed.

Some embodiments according to the invention comprise a non-transitorydata carrier having electronically readable control signals that areable to cooperate with a programmable computer system such that one ofthe methods described herein is performed.

Generally, embodiments of the present invention can be implemented as acomputer program product with a program code, wherein the program codeis effective to preform one of the methods when the computer programproduct is performed on a computer. The program code can be stored, forexample, on a machine readable carrier.

Other embodiments comprise the computer program for preforming one ofthe methods described herein, stored on a machine readable carrier.

In other words, one embodiment of the inventive method is a computerprogram with a program code for executing one of the methods describedherein when the computer program is performed on a computer.

A further embodiment of the inventive method is a data carrier (or adigital memory medium or a computer readable medium) which comprises,recorded on the same, the computer program for performing one of themethods described herein.

Thus, a further embodiment of the inventive method is a data stream or aseries of signals representing the computer program for executing one ofthe methods described herein. The data stream or the series of signalscan be configured to be transmitted via a data communication connection,such as via the Internet.

A further embodiment comprises a processing means, for example acomputer or a programmable logic device which is configured or adaptedto perform one of the methods described herein.

A further embodiment comprises a computer on which the computer programfor performing one of the methods described herein is installed.

In some embodiments, a programmable logic device (for example a fieldprogrammable gate array) can be used to perform one or all of thefunctionalities of the methods described herein. In some embodiments, afield programmable gate array can cooperate with a microprocessor toperform one of the methods described herein. Generally, the methods areperformed by any hardware device.

While this invention has been described in terms of several advantageousembodiments, there are alterations, permutations, and equivalents whichfall within the scope of this invention. It should also be noted thatthere are many alternative ways of implementing the methods andcompositions of the present invention. It is therefore intended that thefollowing appended claims be interpreted as including all suchalterations, permutations, and equivalents as fall within the truespirit and scope of the present invention.

The invention claimed is:
 1. Apparatus for reverberation time extension,comprising: a module for calculating wave field synthesis information, asignal processor for generating a plurality of audio output signals fora plurality of loudspeakers based on a plurality of audio input signalsthat have been recorded by a plurality of microphones, and based on thewave field synthesis information, and an operating unit for determininga virtual position of one or several virtual walls, wherein the modulefor calculating wave field synthesis information is implemented tocalculate the wave field synthesis information based on the virtualposition of the one or several virtual walls, wherein, for at least oneof the virtual walls, the virtual position is adjustable by theoperating unit, and wherein the module for calculating wave fieldsynthesis information is implemented using a hardware apparatus or usinga computer or using a combination of a hardware apparatus and acomputer.
 2. Apparatus according to claim 1, wherein the module forcalculating wave field synthesis information is implemented to calculatedelay values and amplitude factor values as wave field synthesisinformation, wherein the delay value indicates the delay by which one ofthe audio input signals is reproduced at one of the loudspeakers in adelayed manner, and wherein the amplitude factor indicates by whatfactor the amplitude of one of the audio input signals is modified toacquire a modified signal which is output at one of the loudspeakers. 3.Apparatus according to claim 2, wherein the module for calculating wavefield synthesis information is implemented to calculate a delay valueand an amplitude factor for each loudspeaker/virtual wall pair for aspecific time, wherein a loudspeaker/virtual wall pair is a pair of oneof the loudspeakers and one of the virtual walls.
 4. Apparatus accordingto claim 3, wherein the module for calculating wave field synthesisinformation is implemented to calculate the delay value and theamplitude factor value for a loudspeaker/virtual wall pair based on thedistance of the loudspeaker and the virtual wall of theloudspeaker/virtual wall pair.
 5. Apparatus according to claim 3,wherein the module for calculating wave field synthesis information isimplemented to set the delay value of a loudspeaker/virtual wall pairthe greater the greater the distance between the loudspeaker and thevirtual wall is.
 6. Apparatus according to claim 3, wherein the modulefor calculating wave field synthesis information is implemented to setthe amplitude value of a loudspeaker/virtual wall pair the smaller thegreater the distance between the loudspeaker and the virtual wall is. 7.Apparatus according to claim 1, wherein the operating unit isimplemented such that at least one of the virtual walls can be shiftedfrom a first virtual position to a second virtual position, such thatthe virtual wall can be shifted arbitrarily in parallel to its firstposition.
 8. Apparatus according to claim 1, wherein the operating unitis implemented such that at least one of the virtual walls can beshifted from a first virtual position to a second virtual position, suchthat the virtual wall can be shifted arbitrarily in a rotatable mannerwith respect to this first position.
 9. Apparatus according to claim 1,wherein the virtual position for all of the virtual walls is adjustableby the operating unit.
 10. Apparatus according to claim 1, wherein theoperating unit is implemented such that each of the virtual walls can beshifted from a first virtual position to a second virtual position, suchthat each virtual wall can be shifted arbitrarily in parallel and in arotatable manner with respect to its first position.
 11. Apparatusaccording to claim 1, wherein the apparatus further comprises aparametric filter for filtering resonance frequencies.
 12. Method forreverberation time extension, comprising: determining a virtual positionof one or several virtual walls; receiving a plurality of audio inputsignals that have been recorded by a plurality of microphones,calculating wave field synthesis information, and generating a pluralityof output signals for a plurality of loudspeakers based on the audioinput signals and based on the wave field synthesis information, whereinthe wave field synthesis information is calculated based on the virtualposition of the one or several virtual walls, and wherein the virtualposition is adjustable for at least one of the virtual walls.
 13. Anon-transitory computer-readable medium comprising a computer programincluding a program code for performing the method according to claim12, when the computer program runs on a computer.
 14. Electroacousticsystem for reverberation time extension, comprising: a plurality ofmicrophones; an apparatus for reverberation time extension according toclaim 1, and a loudspeaker array comprising a plurality of loudspeakers,wherein the plurality of microphones is implemented to generate aplurality of audio input signals fed into the apparatus forreverberation time extension, and wherein the plurality of loudspeakersof the loudspeaker array are implemented to have the audio outputsignals fed in by the apparatus for reverberation time extension and toreproduce the fed-in audio output signals.
 15. Method for reverberationtime extension by means of an electroacoustic system, comprising:recording a plurality of audio input signals by a plurality ofmicrophones; performing the method for reverberation time extensionaccording to claim 12 for generating a plurality of audio outputsignals, wherein receiving the plurality of audio input signalscomprises that that plurality of audio input signals that have beenrecorded by the plurality of microphones is received, and outputting theplurality of audio output signals by means of a loudspeaker arraycomprising a plurality of loudspeakers.